D.S.P/MATLAB Theory & Experiments

DSP LAB RECORD
CONTENTS
1. INTRODUCTION TO TMS320C6713 DSK ........................................................... 2

2. FEATURES OF C6713 DSK................................................................................... 4
3. INTRODUCTION TO CODE COMPOSER STUDIO ............................................ 6
4. LINEAR & CIRCULAR CONVOLUTION .......................................................... 11
5. SINE WAVE AND SQUARE WAVE ................................................................... 15
6. TMS320C6713 DSK CODEC (TLV320AIC23) CONFIGURATION USING -
BOARD SUPPORT LIBRARY .................................................................................. 19
7. SINE WAVE USING DSK6713 ............................................................................ 21
8. ADVANCED DISCRETE TIME FILTER ( FIR ) ................................................. 23
9. AM FM & PWM USING MATLAB ..................................................................... 27
10. CONVOLUTION .................................................................................................. 31
11. FIR FILTERS......................................................................................................... 33
12. BUTTERWORTH IIR FILTERS .......................................................................... 36
13. CHEBYSHEV TYPE 1 FILTERS ......................................................................... 44
14. CHEBYSHEV TYPE 2 FILTERS ......................................................................... 50
SCT COLLEGE OF ENGINEERING Page 1

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INTRODUCTION TO TMS320C6713 DSK
The C6713 DSK builds on Ti‟s industry-leading line of low cost, easy-to-use DSP
Starter Kit (DSK) development boards. The high performance board features the
TMS320C6713 floating point DSP, capable of performing 1350 million floating point
operations per second(MFLOPS) , the C6713 DSP makes the C6713 DSK the most
powerful DSK development board.
The DSK is USB port interfaced platform that allows efficiently develop and testing
applications for C6713. The DSK consists of a C6713-based printed circuit board that
will serve as a hardware reference design for TI‟s customer‟s product s. With the
extensive host PC and target DSP software support, including bundled TI tools, the
DSK provides ease-of -use and capabilities that are attractive to DSP engineers.
The C6713 DSK has a TMS 320C6713 DSP onboard that allows full speed
verification of the Code Composer Studio. The C6713 DSK provides
 A USB interface
 SDRAM and ROM
 An analog interface circuit for data conversion (AIC)
 An I/O port
 Embedded JTAG emulation support
Connectors on the C6713 DSK provide DSP external memory interface (EMIF) and
peripheral signals that enable its functionality to be expanded with custom or third
party daughter boards.
The DSK provides a C6713 hardware reference design that can assist in the
development of C6713 based products. In addition to providing a reference for
interfacing the DSP to various types of memories and peripheral interfaces, the design
also address power, clock, JTAG and parallel peripheral interfaces.

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The C6713 DSK includes a stereo codec. The analog interface circuit (AIC) has the
following characteristics:
High performance Stereo Codec
 90 dB SNR Multibit Sigma –Delta ADC (A- weighted at 48kHz)

 100 dB SNR Multibit Sigma Delta DAC (A- weighted at 48kHz)
 1.42 V – 3.6 V Core Digital Supply: compatible With TI C54x DSP Core
Voltages
 2.7 V – 3.6 V Buffer and Analog Supply: Compatible with both TI C54x DSP
Buffer Voltages
 8 kHz – 96 KHz Sampling Frequency Support
Software Control via TI McBSP –Compatible Multiprotocol Serial Port
 12 C compatible and SPI – compatible Serial Port Protocols

 Glueless Interface to TI McBSP's
Audio- Data Input/Output via TI McBSP- Compatible Programmable Audio Interface
 12 S- compatible interface requiring only one McBSP for both ADC and DAC
 Standard 12S MSB or LSB justified – Data Transfers
 16/20/24/32- Bit Word Length

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FEATURES OF C6713 DSK
The 6713 DSK is a low-cost standalone development platform that enables customers
to evaluate and develop applications for the TI C67XX DSP family. The DSP also
serves as a hardware reference design for the TMS320C6713 DSP. Schematics, logic
equations and application notes are available to ease hardware development and
reduce time to market.
The DSK uses the 32bit EMIF for the SDRAM (CE0) and daughtercard expansion
interface ( CE2 and CE3). The Flash is attached to CE1 of the EMIF in 8-bit mode.
An on-board AIC23 codec allows the DSP to transmit and receive analog signals.
McBSP0 is used for the codec control interface and McBSP1 is used for data. Analog
audio I/O is done through four 3.5mm audio jacks that correspond to microphone
input, line input, line output and headphone output. The codec can select the
microphone or line input as the active input. The analog output is driven to both the
line out (fixed gain) and headphone (adjustable gain) connectors. McBSP1 can be re-
routed to the expansion connectors in software.
A programmable logic device called CPLD is used to implement glue logic that ties
the board components together. The CPLD has a register based user interface the lets
the user configure the board by reading and writing to the CPLD registers. The
registers reside at the midpoint of CE1.
The DSK include 4 LEDs and 4 DIP switches as a simple way to provide the user
with interactive feedback. Both are accessed by the reading and writing to the CPLD
register.
An included 5V external power supply is used to power is used to power the board.
On-board voltage regulators provide 1.26V DSP core voltage, 3.3V digital and 3.3V
analog voltages. A voltage supervisor monitors the internally generated voltage and
will hold the board in reset until the supplies are within operating specifications and
the reset button is released. If desired, JP1 and JP2 can be used as power test points
the core and I/O power supplies.
Code Composer communicates with the DSP through an embedded JTAG emulator
with a USB host interface. The DSK can also be used with an external emulator
through the external JTAG connector.

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INTRODUCTION TO CODE COMPOSER STUDIO
Code Composer is the DSP industry‟s first fully integrated development environment
(IDE) with DSP- specific functionality. With a familiar environment like MS- based
C++, Code composer lets you edit, build, debug, profile and manage projects from a
single unified environment. Other unique features include graphical signal analysis,
injection/extraction of data signals via file I/O, multi-processor debugging, automated
testing and customization via a C-interpretive scripting languages and much more.
CODE COMPOSER FEATURE INCLUDE:
 IDE
 Debug IDE
 Advanced watch windows
 Integrated editor
 File I/O, probe points and graphical algorithm scope probes
 Advanced graphical signal analysis
 Interactive profiling
 Automated testing and customization via scripting
 Visual project management system
 Compile in the background while editing and debugging
 Multi-processor debugging
 Help on the target DSP

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PROCEDURE TO WORK ON CODE COMPOSER STUDIO
1. To create a new project
Project  New
2. To create source file
File  New  Source file

Save the source file in project folder
File  Save

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3. To add source file to the project
Project  Add files to project  <source file>
4. To add rts6700.lib & hello.cmd
Project  Add files to project  rts6700.lib

Path : C:\CCstudio\c6000\lib\rts6700.lib
Note: Select Object & library files (*.o, *.l) in Type of file
Project  Add files to project  hello.cmd

Path: C:\CCstudio\tutorial\dsk6713\hello1\hello.cmd
Note: Select Linker command file (*.cmd) in Type of file

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5. To compile
Project  Compile file ( or use icon or ctrl+F7 )
6. To build or link
Project build (or use F7 )

(which will create a .out file in project folder)
7. To load the program:
File  Load Program  <select the .out file in debug folder in project folder>
8. To run the program
Debug  Run ( or use icon or F5)

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To perform Single Step Debugging:
1. Keep the cursor on the line from point to start single step debugging.
To set the break point click icon from tool bar menu.
2. Load the .out file onto the target
3. Go to View and select Watch window
4. Select Debug  Run (F5)
5. Execution should halt at break point
6. Now press F10. See the changes happening the watch window
7. Similarly go to view & select CPU registers to view the changes happening in
CPU registers.
8. Repeat steps 2 to 6

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Experiment no 1
LINEAR & CIRCULAR CONVOLUTION
AIM
To implement circular and linear convolution using Code Composer Studio.
PROCEDURE
1. Open Code Composer Setup and select C6713 simulator, click save and quit
2. Start a new project using „Project  New‟ pull down menu, save it in a
separate directory (D:\My projects) with file name lconv.pjt
3. Create a new source file using File  New  Source file menu and save it in
the project folder(linear.c)
4. Add the source file (linear.c) to the project
Project  Add files to Project  Select linear.c
5. Add the linker command file hello.cmd
Project  Add files to Project
(path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)
6. Add the run time support library file rts6700.lib
(path: C\CCStudio\cgtools\lib\rts6700.lib)
7. Compile the program using „project  Compile‟ menu or by Ctrl+F7
8. Build the program using „project  Build‟ menu or by F7
9. Load the linear.out file (from project folder lcconv\Debug) using
File  Load Program
10. Run the program using „Debug  Run‟ or F5
11. To view the output graphically
Select View  Graph  Time and Frequency
12. Repeat the steps 2 to 11 for circular convolution
PROGRAM
Linear Convolution
#include<stdio.h>
#define LENGTH1 6
#define LENGTH2 4
int x[LENGTH1+LENGTH2-1]={1,2,3,4,5,6,0,0,0};

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int h[LENGTH1+LENGTH2-1]={1,2,3,4,0,0,0,0,0};
int y[LENGTH1+LENGTH2-1];
main()
{
int i=0,j;
for(i=0;i<LENGTH1 + LENGTH2-1;i++)
{
y[i]=0;
for(j=0;j<=i;j++)
y[i]+=x[j]*h[i-j];
}
for(i=0;i<LENGTH1 + LENGTH2-1;i++)
printf("%d\n",y[i]);
}
RESULT
The convoluted sequence is obtained as
1 4 10 20 30 40 43 38 24

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Circular convolution
#include<stdio.h>
int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30];
void main()
{
printf("Enter the length of 1st sequence\n");
scanf("%d",&m);
printf("Enter the length of 2nd sequence\n");
scanf("%d",&n);
printf("Enter the 1st sequence\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf("Enter the 2nd sequence\n");
for(j=0;j<n;j++)
scanf("%d",&h[j]);
if(m-n!=0)
{
if(m>n)
{
for(i=n;i<m;i++)
h(i)=0;
n=m;
}
for(i=m;i<n;i++)
x[i]=0;
m=n;
}
y[0]=0;
a[0]=h[0];
for(j=1;j<n;j++)
a[j]=h[n-j];
for(i=0;i<n;i++)
y[0]+=x[i]*a[i];

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for(k=1;k<n;k++)
{
y[k]=0;
for(j=1;j<n;j++)
x2[j]=a[j-1];
x2[0]=a[n-1];
for(i=0;i<n;i++)
{
a[i]=x2[i];
y[k]+=x[i]*x2[i];
}
}
printf("The circular convolution is \n");
for(i=0;i<n;i++)
printf("%d\t",y[i]);
}
RESULT
Enter the length of 1st sequence 4
Enter the length of 2nd sequence 4
Enter the 1st sequence 3 2 1 4
Enter the 2nd sequence 1 2 3 4
The circular convolution is 22 24 30 24

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Experiment 2
SINE WAVE AND SQUARE WAVE
AIM
To generate a sine wave and square wave using C6713 simulator
PROCEDURE
1. Open Code Composer Setup and select C6713 simulator, click save and quit
2. Start a new project using „Project  New‟ pull down menu, save it in a
separate directory (D:\My projects) with file name sinewave.pjt
3. Create a new source file using File  New  Source file menu and save it in
the project folder(sinewave.c)
4. Add the source file (sinewave.c) to the project
Project  Add files to Project  Select sinewave.c
5. Add the linker command file hello.cmd
(path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)
6. Add the run time support library file rts6700.lib
(path: C\CCStudio\cgtools\lib\rts6700.lib)
7. Compile the program using „project  Compile‟ menu or by Ctrl+F7
8. Build the program using „project  Build‟ menu or by F7
9. Load the sinewave.out file (from project folder lcconv\Debug) using
File  Load Program
10. Run the program using „Debug  Run‟ or F5
11. To view the output graphically
Select View  Graph  Time and Frequency
12. Repeat the steps 2 to 11 for square wave
PROGRAM
Sine Wave
#include <stdio.h>
#include <math.h>
float a[500];
void main()
{
int i=0;
for(i=0;i<500;i++)

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{
a[i]=sin(2*3.14*10000*i);
}
}
Square wave
#include <stdio.h>
#include <math.h>
int a[1000];
void main()
{
int i,j=0;
int b=5;
for(i=0;i<10;i++)
{ for (j=0;j<=50;j++)
{
a[(50*i)+j]=b;
}
b=b*(-1) ;
}
}

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RESULT
The sine wave and square wave has been obtained.
Sine Wave

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Square Wave:

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Experiment 3
TMS320C6713 DSK CODEC (TLV320AIC23)

CONFIGURATION USING BOARD SUPPORT LIBRARY
AIM
To configure the codec TLV320AIC23 for a talk through program using the board
support library
PREREQUISITES
TMS320C6713 DSP starter kit, PC with Code Composer Studio, CRO, Audio source,
Speakers, Signal Generator.
PROCEDURE
1. Connect Speaker to the LINE OUT socket.

2. Connect the line out from the PC to the LINE IN socket.
3. Now switch ON the DSK and bring up Code Composer Studio on PC
4. Create a new project with name codec.pjt
5. From File menu  New  DSP/BIOS Configuration  Select dsk6713.cdb
and save it as “xyz.cdb”
6. Add xyz.cdb to the current project
7. Create a new source file and save it as codec.c
8. Add the source file codec.c to the project
9. Add the library file “dsk6713bsl.lib” to the project
( Path: C:\CCStudio\C6000\dsk6713\lib\dsk6713bsl.lib)
10. Copy files “dsk6713.h” and “dsk6713_aic23.h” to the Project folder
11. Build (F7) and load the program to the DSP Chip ( File  Load Program)
12. Run the program (F5)
13. Give an audio output from the PC and notice the output in the speaker
14. Vary the sampling frequency using the DSK6713_AIC23_SetFreq
PROGRAM
Codec.c
#include"xyzcfg.h"
#include"dsk6713.h"
#include"dsk6713_aic23.h"

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DSK6713_AIC23_Config config= DSK6713_AIC23_DEFAULTCONFIG;

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
int l_input,r_input,l_output,r_output;
DSK6713_init();
hCodec=DSK6713_AIC23_openCodec(0,&config);
DSK6713_AIC23_setFreq= DSK6713_AIC23_FREQ_48KHZ;
while(1)
{
while(!DSK6713_AIC23_read(hCodec,&l_input));
while(!DSK6713_AIC23_read(hCodec,&r_input));
r_output = r_input;
l_output = l_input;
while(!DSK6713_AIC23_write(hCodec,l_output));
while(!DSK6713_AIC23_write(hCodec,r_output));
}
DSK6713_AIC23_closeCodec(hCodec);
}
RESULT
The Codec TMS320AIC23 Successfully configured using the board support library
and output is verified.

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Experiment 4
SINE WAVE USING DSK6713
AIM
To generate a real time sine wave using TMS320C6713 DSK
PROCEDURE

3. Create a new project with name sinewave.pjt
6. Create a new source file and save it as sinewave.c
7. Add the source file sinewave.c to the project
PROGRAM
Sinewave.c
#include "xyzcfg.h"
#include "dsk6713.h"
#include "dsk6713_aic23.h"
#include <stdio.h>
#include <math.h>
float a[500],b;
DSK6713_AIC23_Config config= DSK6713_AIC23_DEFAULTCONFIG;
void main()
{

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int i=0;
Int l_output,r_output;
DSK6713_init();
hCodec=DSK6713_AIC23_openCodec(0,&config);
DSK6713_AIC23_setFreq=DSK6713_AIC23_FREQ_48KHZ;
for(i=0;i<500;i++)
{
a[i]=sin(2*3.14*10000*i);
}
while(1)
{
for(i=0;i<500;i++)
{
b=400*a[i];
while(!DSK6713_AIC23_write(hCodec,b));
}
}
}
RESULT
The sine wave tone have been successfully generated and obtained through speaker

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Experiment 5
ADVANCED DISCRETE TIME FILTER ( FIR )
AIM
To generate filter coefficients using MATLAB and to implement Audio filter using
DSK6713
PROCEDURE

2. Connect the line out from the PC to the LINE IN socket.
4. Create a new project with name firfilter.pjt
7. Create a new source file and save it as firfilter.c
8. Add the source file firfilter.c to the project

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Procedure for Generating Filter Coefficients in MATLAB
1. Open Matlab
2. Start  Toolboxes  Filter Design  Filter Design & Analysis Tool
3. Select the filter type and give the order and specifications for the filter and
click Design filter

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4. From the File option select Export (or use ctrl + E)
5. Select export to Coefficient File(ASCII) and save the coefficients as txt file
6. Use these coefficients to form the coefficients in the C-file.
PROGRAM
#include"xyzcfg.h"
#include"dsk6713.h"
#include"dsk6713_aic23.h"
#define FIR_ORD 20
short fir_filter(Uint32 in,short *in_buff,float *coeff);
float fir_coeff[FIR_ORD+1]={ -0.03076334540056, -0.005410850195565,
0.03739961263014, 0.02059285910535, -0.04315598597689, -0.04556930460411,
0.04761022772871, 0.09663870571612, -0.05042696667111, -0.3229392723481,
0.5652971907268, -0.3229392723481, -0.05042696667111, 0.09663870571612,
0.04761022772871, -0.04556930460411, -0.04315598597689, 0.02059285910535,
0.03739961263014, -0.005410850195565, -0.03076334540056 };

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short x_buff[FIR_ORD+1];
DSK6713_AIC23_Config config = DSK6713_AIC23_DEFAULTCONFIG;
void main()
{
Uint32 l_input,r_input,l_output,r_output;
DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,&config);
DSK6713_AIC23_setFreq(hCodec,DSK6713_AIC23_FREQ_48KHZ);
while(1)
{
while(!DSK6713_AIC23_read(hCodec,&l_input));
while(!DSK6713_AIC23_read(hCodec,&r_input));
l_output=5*fir_filter(l_input,x_buff,fir_coeff);
r_output=5*fir_filter(l_input,x_buff,fir_coeff);
while(!DSK6713_AIC23_write(hCodec,l_output));
while(!DSK6713_AIC23_write(hCodec,r_output));
}
}
short fir_filter(Uint32 in, short *in_buff, float *coeff)
{
int out=0;
int i;
for(i=FIR_ORD;i>0;i--)
{
in_buff[i]=in_buff[i-1];
out=out+in_buff[i]*coeff[i];
}
in_buff[0]=in;
out=out+in_buff[0]*coeff[0];
return out;
}
RESULT
The FIR filter has been implemented and the output is verified

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Experiment 6
AM FM & PWM USING MATLAB
AIM
To implement a program in MATLAB to generate AM, FM and PWM wave.
PROGRAM
Amplitude Modulated wave

clc; clear all; close all;
t = 0:0.0001:0.015;
m = input(„Modulation Index = ‟);
fm=input(„Frequency of input wave=‟);
fc=input(„Frequency of carrier wave=‟);
x=sin(2*pi*fm*t);
y=sin(2*pi*fc*t);
e=(1+(m.*x)).*y;
plot(e);
title(„AM WAVE‟);
xlabel(„TIME INDEX‟);
ylabel(„AMPLITUDE‟);
RESULT
Modulation Index = 0.5

Frequency of input wave= 100
Frequency of carrier wave= 1000

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Frequency modulated wave

clc; close all; clear all;
a= input („enter amplitude‟);
fc=input(„enter carrier freq‟);
fm=input(„enter modulation freq‟);
m=input(„enter modulation index‟);
n=0:0.001:.2;
y=a*sin(2*pi*fc*n-(m*cos(2*pi*fm*n)));
plot(n,y);
title(„FM‟);
xlabel(„TIME‟);
ylabel(„AMPLITUDE‟);

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RESULT
enter amplitude = 1
enter carrier freq =100
enter modulation freq =10
enter modulation index =5

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Pulse Width Modulated Wave

fs=100;
t= 0:1/(5*fs):2;
x=sawtooth(2*pi*20*t);
m=0.75*sin(2*pi*t);
k=length(x);
for i=1:k
if (m(i)>=x(i))
pwm(i)=1;
else if (m(i)<x(i)
pwm(i)=0;
end;
end;
subplot(2,2,3);
plot(t,pwm,t,m);
axis([0,0.5,-2,2]);
RESULT

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Experiment 7
CONVOLUTION
AIM
To write a program in MATLAB for convolution of two sequence.
PROGRAM

a=input(„enter first sequence ‟);
b= input(„enter second sequence ‟);
c= fliplr(b);
d=length(a);
e=length(b);
f=[zeros(1,e-1),a,zeros(1,e)];
g=0;
i=d+e-1;
for h=0:(d+e-1);
j=[zeros(1,g),c,zeros(1,i)];
k=sum(f.*j);
con(g+1)=k;
g=g+1;
i= i-1;
end
m= 0:d+e-1;
stem(m,con);
title(„convoluted sequence‟);
xlabel(„time‟);
ylabel(„amplitude‟);

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RESULT
enter first sequence [1 2 3 4]

enter second sequence [4 3 2 1 ]

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Experiment 8
FIR FILTERS
AIM
To write a program in MATLAB for implementing FIR filters.
PROGRAM

rp= input('Enter the passband ripple ');
rs= input('Enter the stop band ripple ');
fp= input('Enter the passband frequency ');
fs=input('Enter the stopband frequency ');
f=input('Enter the sampling frequency ');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=boxcar(n1);
%LOW PASS FILTER
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);

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plot(o/pi,m);
title('1. LPF');
ylabel('Gain(dB)');
xlabel(' Normalised Frequency');
%HIGH PASS FILTER
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
title('2. HPF');
ylabel('Gain(dB)');
%BAND PASS FILTER
wn=[wp,ws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
title('3. BPF');
ylabel('Gain(dB)');
%BAND STOP FILTER
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);

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title('4. BSF');
ylabel('Gain(dB)');
RESULT
Enter the passband ripple .05

Enter the stopband ripple .04
Enter the passband frequency 1500
Enter the stopband frequency 2000
Enter the sampling frequency 9000

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Experiment 9
BUTTERWORTH IIR FILTERS
AIM
To design and simulate Butterworth IIR filters using MATLAB
PROGRAM
Low Pass Filter

wp= input('Enter the passband frequency ');
ws=input('Enter the stopband frequency ');
fs=input('Enter the sampling frequency ');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs);
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('IIR LPF');
ylabel('Gain(dB)');
xlabel('Normalised frequency');
subplot(2,1,2);

DSP LAB RECORD
plot(om/pi,an);
ylabel('Phase(radians)');
RESULT

Enter the stop band ripple 50

DSP LAB RECORD
HPF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=butter(n,wn,'high');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('IIR HPF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
BPF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=butter(n,wn);
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('IIR BPF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT

Enter the passband frequency [800 1200]
Enter the stopband frequency [500 1500]

DSP LAB RECORD
BSF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=butter(n,wn,'stop');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('IIR BSF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
Experiment 10
CHEBYSHEV TYPE 1 FILTERS
AIM
To design and simulate Chebyshev type 1 filters using MATLAB
PROGRAM
LPF

w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn);
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 1 LPF');
ylabel('Gain(dB)');
subplot(2,1,2);

DSP LAB RECORD
plot(om/pi,an);
RESULT


DSP LAB RECORD
HPF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=cheby1(n,rp,wn,'high');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 1 HPF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
BSF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=cheby1(n,rp,wn,'stop');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 1 BSF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
Experiment 11
CHEBYSHEV TYPE 2 FILTERS
AIM
To design and simulate Chebyshev type 2 filters using MATLAB
PROGRAM
LPF
w1=2*wp/fs;
w2=2*ws/fs;
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 2 LPF');
ylabel('Gain(dB)');

DSP LAB RECORD
subplot(2,1,2);
plot(om/pi,an);
RESULT

DSP LAB RECORD
HPF

w1=2*wp/fs;
w2=2*ws/fs;
[b,a]=cheby2(n,rp,wn,'high');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 2 HPF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
BPF

w1=2*wp/fs;
w2=2*ws/fs;
[n wn]=cheb2ord(w1,w2,rp,rs);
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 2 BPF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


DSP LAB RECORD
BSF

w1=2*wp/fs;
w2=2*ws/fs;
[n wn]=cheb2ord(w1,w2,rp,rs);
[b,a]=cheby2(n,rp,wn,'stop');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('Chebyshev Type 2 BSF');
ylabel('Gain(dB)');
subplot(2,1,2);
plot(om/pi,an);

DSP LAB RECORD
RESULT


D.S.P/MATLAB Theory &amp; Experiments

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DSP LAB RECORD

1. INTRODUCTION TO TMS320C6713 DSK ........................................................... 2

SCT COLLEGE OF ENGINEERING Page 1

INTRODUCTION TO TMS320C6713 DSK

SCT COLLEGE OF ENGINEERING Page 2

High performance Stereo Codec

 90 dB SNR Multibit Sigma –Delta ADC (A- weighted at 48kHz)

Software Control via TI McBSP –Compatible Multiprotocol Serial Port

 12 C compatible and SPI – compatible Serial Port Protocols

Audio- Data Input/Output via TI McBSP- Compatible Programmable Audio Interface

SCT COLLEGE OF ENGINEERING Page 3

FEATURES OF C6713 DSK

SCT COLLEGE OF ENGINEERING Page 4

SCT COLLEGE OF ENGINEERING Page 5

INTRODUCTION TO CODE COMPOSER STUDIO

CODE COMPOSER FEATURE INCLUDE:

SCT COLLEGE OF ENGINEERING Page 6

PROCEDURE TO WORK ON CODE COMPOSER STUDIO

1. To create a new project

2. To create source file

File  New  Source file

SCT COLLEGE OF ENGINEERING Page 7

3. To add source file to the project

Project  Add files to project  <source file>

4. To add rts6700.lib & hello.cmd

Project  Add files to project  rts6700.lib

Project  Add files to project  hello.cmd

SCT COLLEGE OF ENGINEERING Page 8

Project build (or use F7 )

7. To load the program:

8. To run the program

Debug  Run ( or use icon or F5)

SCT COLLEGE OF ENGINEERING Page 9

To perform Single Step Debugging:

2. Load the .out file onto the target

3. Go to View and select Watch window

4. Select Debug  Run (F5)

5. Execution should halt at break point

SCT COLLEGE OF ENGINEERING Page 10

LINEAR & CIRCULAR CONVOLUTION

To implement circular and linear convolution using Code Composer Studio.

SCT COLLEGE OF ENGINEERING Page 11

The convoluted sequence is obtained as

SCT COLLEGE OF ENGINEERING Page 12

SCT COLLEGE OF ENGINEERING Page 13

Enter the length of 1st sequence 4

Enter the length of 2nd sequence 4

Enter the 1st sequence 3 2 1 4

Enter the 2nd sequence 1 2 3 4

The circular convolution is 22 24 30 24

SCT COLLEGE OF ENGINEERING Page 14

SINE WAVE AND SQUARE WAVE

To generate a sine wave and square wave using C6713 simulator

SCT COLLEGE OF ENGINEERING Page 15

SCT COLLEGE OF ENGINEERING Page 16

The sine wave and square wave has been obtained.

SCT COLLEGE OF ENGINEERING Page 17

SCT COLLEGE OF ENGINEERING Page 18

TMS320C6713 DSK CODEC (TLV320AIC23)

1. Connect Speaker to the LINE OUT socket.

SCT COLLEGE OF ENGINEERING Page 19

DSK6713_AIC23_Config config= DSK6713_AIC23_DEFAULTCONFIG;

SCT COLLEGE OF ENGINEERING Page 20

D.S.P/MATLAB Theory & Experiments

D.S.P/MATLAB Theory & Experiments